similar to: Ztdummy - is it requirement?

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Few weeks back local telco introduced option of custom ring tones. I am not talking about your phone ring tones but about ring tones you hear in your headset while phone is ringing on the other end. If I understand correctly, ringing tone is generated localy on asterisk if you are connected to phone network with E1/T1 connection. Which means that instead of regular beep-beep tone we could send

OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem?

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

Hi, I've found that neither Michael Manousos patch nor ztdummy driver do not fix musiconhold sound interruption problem up to acceptable quality level. Sound is choppy here anyway. It is my understanding (please correct me if I'm wrong) that if I have a Digium card in my asterisk machine, these problems should be gone 'cause those cards provide some reliable timing. So I have no

Can someone help me what went wrong with this call? This call was initiated from dev/ttyI0 device on my asterisk server to mediatrix unit. Mediatrix unit user received the call and call started. I can hear them OK but they can not hear me correctly (cut-off sound, noise). Call was finally hunged up. Can anyone point out if there was something wrong? -*CLI> sip debug SIP Debugging Enabled

I am trying to music on hold but I am having all sorts of problems with it. I am running RH9 and the latest version of Asterisk as of yesterday. Here is what I did to test it: 1. I manually deleted the mpg123 softlink to mpg321. 2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed the the archive and loaded ztdummy.o module. 3. I threw a couple extra files in the mohmp3 directory

Any idea what causes the warning "Unable to open pseudo channel for timing... Sound may be choppy."? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated!

I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when

Hello, everyone I'm relatively new to Asterisk (and VOIP in general), but I have a project that it will really help with. So, I setup a test system on an ancient 400MHz P3 we had lying around. It worked great. I had a test dialplan working, and had no trouble connecting to it with SIP using 3CX SoftPhone over our LAN (and over the Net through our NAT). So, we went ahead and bought a

Hi, I've install zaptel and I don't have a Digium card installed in the machine. So I want to install ztdummy to have Music On Hold working. I've follow these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and at the point "modprobe ztdummy" it's failing. I'm getting these messages: root@dubic:~# modprobe ztdummy FATAL: Error inserting ztdummy

Hi i followed this instructions for installing ztdummy on a 2.6 kernel (taken from http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy ) cd /usr/src/zaptel * READ /usr/src/zaptel/README.udev and follow the steps * check modules on: /etc/sysconfig/zaptel> if you have no digium hardware comment out all modeules except ztdummy. - make linux26 - make install -

Hi there, anyone has a suggestion about what do I need to do to solve this? Any comments appreciated. Philipp [root@asterisk]# modprobe ztdummy /lib/modules/2.4.22/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rd2dbbfe0 /lib/modules/2.4.22/misc/zaptel.o: unresolved symbol ppp_input_error_Rd2ca1e76 /lib/modules/2.4.22/misc/zaptel.o: unresolved symbol ppp_unregister_channel_Rb8cf9a74

Hi! When I try to load the ztdummy driver via "insmod ztdummy", I get the following errors: /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_register I'm using zaptel-1.0.8 and

Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I

Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn lines in either Loop Start or Ground Start mode, handles incoming CallerID, and generates either Dial Tone (back towards the

Hi all, Trying to build ztdummy on an old redhat 7.3 box running kernel 2.4.20-43.7.legacysmp. Yes, I have the kernel sources installed. Yes, I set them up with make oldconfig; make dep. The build error is: make ztdummy gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/

Hi to everybody, seems that I cannot load the zaptel modules: ztdummy says the following: [root@Hayabusa misc]# modprobe ztdummy /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_register

Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the